Monday, December 31, 2012

Enjoy our new release of"Solace" music album by Jordan
Listen to a sample or download here
View "Solace" sample trailer here
Purchase a CD hereComing Soon to Pandora Internet Radio!! "Like" us on Facebook.

From all of us here at No Limit Sound Productions, we would like thank you for being with us and would like to you wish you all a

Try close-miking a flute or any other woodwind instrument
and you will pick up a whole load of key clicking. What can you do? How
much is too much?

We received a recording of solo flute at RP Towers
recently. The sender was enquiring whether the amount of key click noise
was too much.

Originally these instruments had no keys, and
therefore no key click noise. That however was more than a hundred years
before the invention of recording so there was no-one around to
appreciate the benefit.

But wily inventors added first one key,
then another, all in the interests of player convenience and extending
the range of the instrument. In modern instruments, all or nearly all
notes are produced using keys, rather than stopping the holes directly
with the fingers.

Inevitably this causes a mechanical noise that is easily picked up by sensitive microphones. So what can you do?

The
first thing you can do as a recording engineer is consider where the
normal listening position for a woodwind instrument would be. Would it
be ten meters away or more if the instrument were in an orchestra? Would
it be two or three meters if you were sitting in the front row in a
small recital? Would it be a couple of centimeters from the instrument
if you wanted to get the same experience as a close mic?

For some
reason that acoustic science finds it difficult to explain, key click
noise seems to disappear with distance. You would almost never become
aware of it in an orchestral concert.

But the close mic sound is very attractive. If you must have that sound, what can you do about the key clicks?

The
first thing to do is have the instrument serviced. Players tend not to
notice the gradual degradation in performance of an instrument between
services. They hear the key clicks every time they practice and cease to
notice them.

This might not be practical and you have to deal with the instrument as it is.

You
could try asking the player not to click so much. This will probably
produce no noticeable benefit and the performance will suffer because
the player is now distracted.

One thing you can do successfully is
move the microphone towards the source of musical sound and away from
the mechanism causing the clicks.

Woodwind instruments produce
sound from three places - the mouthpiece or reed, the opening at the end
(the bell in the clarinet and oboe), and from any holes in the
instrument that are not covered. This last source changes from note to
note, and you will hear this clearly if you listen from close range.

So
if you move the microphone closer to the mouthpiece or reed, or to the
opening at the end, you will increase the amount of musical sound
slightly and decrease the level of the clicks. You can also experiment
moving the microphone around the instrument to the side that has fewer
mechanical parts.

Two points...

1. Don't expect miracles. Expect a small improvement at best.
2. Moving the mic will change the sound quality, perhaps not for the better.

There
is however another solution that trumps all the rest, and that is to
accept these noises as part of the texture of the instrument.

All acoustic instruments have texture and this is something that gives an instrument its character.

This
can come as a surprise if you have learnt recording on synthesized and
sampled instruments. But texture can be a large part of the 'soul' of a
recording and something to be welcomed, in reasonable quantities.

We would like to hear about your experiences with the 'texture' of acoustic instruments.

What non-musical sounds do you like? And what drives you crazy?

If you can e-mail us an example, we would love to hear it and perhaps publish it on the site - newsletter@recordproducer.com

Friday, December 28, 2012

If an offer seems too good to be true, then it's probably a scam. Would you fall for this one?

If I had a dollar, or better still a UK pound, for every
time I've won the Nigerian National Lottery, then I would have as much
money as if I had won a real lottery. Weight loss, hair loss, loss of
sexual function - there's a cheap cure for just about every medical
condition there is. According to my e-mail inbox!

And then there are the phishing attacks. There's not a day goes by
without some bank that I've never dealt with inviting me to 'update my
account details'. Occasionally I get one from what seems to be my bank,
but of course it isn't. It's just someone who wants to trick me into
giving them my password.

But I was shocked the other day to
receive a phishing attack email in the name of Waves, the renowned
developers of high quality, but expensive, plug-ins. Apparently I could
buy Waves plug-ins for as little as $18. Wow, if that were true it would
be amazing!

What shocked me was that I thought that phishing attacks only
involved banks and other institutions where account information could be
valuable to a criminal. But Waves? How could anyone make much money
from knowing people's Waves account details? Well I'm not a criminal
mastermind so I wouldn't know. (We don't need ideas, thank you.)

(It did occur to me that it might be fake software, but I never seem
to get e-mails for any kind of fake software, so I guess there isn't
much of a profit in it.)

What worried me further was the thought that this kind of attack
could spread through other audio manufacturers and software developers.
Since this is an area I need to keep in touch with, it would make my
life a nightmare.

Since I am hardened, through experience, to phishing attacks I know
what to look for. One key piece of information is shown in the status
bar below the message when I hover on a link - does the status bar show
the correct website? If it doesn't, then to me that could suggest that
the email is a phishing attack.

And lo and behold... the links in the Waves email go to trailer.web-view.net

So I typed in trailer.web-view.net into my browser, which got me,
"403 - Forbidden: Access is denied". This was not looking good and I
felt even more strongly that this is a scam, and clicking any link would
invite trouble of all kinds.

Except...

It isn't a scam at all. It is a genuine offer from Waves. I didn't click on the link but instead typed www.waves.com into my browser's address bar, which is good anti-phishing practice.

You can indeed buy the Renaissance EQ for $38, and AudioTrack for a mere $18!

Clearly
someone has made a big mistake here because the e-mail showed every
sign of being a phishing attempt. But if you want to grab yourself a
bargain, now is the time to type www.waves.com into your browser.

P.S.
If anyone has an interesting e-mail scam to report, I'm sure we would
all benefit from hearing of it. I'm afraid I did once fall for 'Someone
has left negative feedback for you on eBay'. Fortunately I realized very
quickly and changed my password. Could have been nasty though.

Is it possible to record a high-quality sounding drum track
in a small space? Or am I just wasting my time recording in a room with a
low ceiling? I am using high quality microphones (worth a few thousand
collectively), a good drum-set with very high end Zildjian cymbals and I
know how to properly mix a kit. Nor am I ignorant of phase problems and
how to deal with them. Nonetheless I have never been able to achieve a
drum sound in a small room that sounded any better than just 'good'. Is
it possible? What should I do?

From time to time we receive a question that is really difficult to answer, and this is one of them.

It
would be nice if we could give a magical solution that would allow
drums to be recorded to a professional standard in a small room, but the
best we can do is offer ideas that will help you progress in a
professional direction, if not get all the way there.

The first
thing to realize is that the sound of professionally-recorded drums is
the sound of professionally-recorded drums in a large room, not a small room. At least five or six meters in the smaller horizontal direction.

And by 'large room' we also mean 'large room with a high ceiling', preferably four meters or more.

Such a room is way beyond what you would normally find in anyone's house or apartment.

It
also helps for the large room to have good acoustics. And of course pro
studios are professionally acoustically designed, so this is what you
would expect.

'Good acoustics' means having a controlled
reverberation. The reverb field should be diffused with many weak
reflections rather than few strong ones. There should not be too much
reverb, and its frequency response should be smooth. I say 'smooth'
rather than 'flat' because it is normal to have a longer reverberation
time at low frequencies rather than high, because low frequencies are
less easily absorbed. This is what we find in rooms all the time, and
the human ear tends to like what it expects.

Small rooms tend to show the effects of standing waves too, where certain frequencies are highly emphasized compared to others.

What
we consider to be good acoustics for recording also implies that the
direct sound and reverberation are separated somewhat in time. So
imagine a single drum strike. In a large room the direct sound will
travel several meters to the nearest surface, other than the floor, then
reflect. In a small room, it will travel over a much shorter distance
and the reverb will build up much more quickly.

So the problems with normal domestic rooms are these...

A few strong reflections rather than many weak ones

Uncontrolled frequency response of the reverberation

Standing waves

Inadequate separation in time between the direct and reverberant sound

Although
it will be impossible to turn a small room acoustically into a large
one, you might consider at least making some improvements.

So you
could use acoustic treatment to fix any standing wave problems. Normally
this is done with tuned panel absorbers or Helmholtz resonators.

You
could control and diffuse the reverb at the same time with suitable
acoustic treatment. Oddly enough, the installation of bookshelves (with
books!) can work well since they are partially absorbent and have
irregular surfaces that break up what reflections that remain.

Controlling
the frequency response of the reverberation normally implies giving
special attention to low frequencies, since these are the most difficult
to absorb. If only porous absorption is used, then only high-mid and
high frequencies will be absorbed leading to an unpleasant imbalance.
Panel absorbers are used for the lows because they take up less space
than porous absorbers and can be tuned to the required range of
frequencies.

Having done all that, what remains is that your room
is small, and the reflections occur within a short time of the direct
sound. You can never prevent that, but what you can consider is to make
the room less reverberant altogether, which means having more
absorption. Since the problem is not in the drums or cymbals but in the
reverberation, then it makes sense to eliminate what is causing the
problem. Perhaps not completely, but enough to make a difference. Since
the quality of plug-in reverbs is excellent these days, you can always
add any ambience you consider to be missing.

If you consider all
of the above carefully, then you will see that making improvements is
going to take an awful lot of hard work.

There is an old English proverb... "If at first you don't succeed, try, try again."

A
more useful version adds at the end, "And if you have tried and tried
again, it's really time to give up and find something more practical!"

It
may indeed be more practical to find somewhere else to record your
drums, or hire a professional studio to record drum tracks.

A RecordProducer.com reader has a song that is too long. How can he edit it so that it is shorter?

"I would like some advice on editing. I have to cut a song from
7:45 to 4:00 because it is too long. Where should I start the editing
process?"

Too long for what?

My first question here would be, "Too long for what?" If this is a
song for commercial release then it is certainly too long. If you are
going to write a song longer than Bohemian Rhapsody then it would really have to be something rather special.

If this is the purpose of the editing, then you need to think about
what is it about the song that will make people buy it? All you have to
do then is cut out the rest.

For instance, it might be that some of the duration is taken up by
chorus repeats at the end. You could have the chorus once at the end and
fade quickly at the start of the first repeat.

Some of the duration might be taken up by an instrumental solo. It's a
rare solo that is the 'hook' of the song and the reason for purchase.
Cut it out, throw it on the studio floor and trample it to death.

Some songs are burdened by too much introduction. Perhaps you can
start straight in with the verse, or at least cut the introduction down.

If after all of these cuts the song is still too long, then you're
going to have to lose a verse or two. Choose the verse or verses that
say the least; that add the least value to a potential purchase.

Cutting for TV

It may be however that the song is to be used as part of a soundtrack
for some kind of TV usage. For a commercial, trailer, program intro or
drama soundtrack. In this case you have much more freedom. You should
identify exactly what it is that makes this song unique; the reason why
the producer chose it. You can now cut out pretty much everything else,
and perhaps even repeat the good bits.

In this situation it is often not a good idea to try and keep a bit
of each section of the song. For example, there is little point in
keeping four bars of the middle eight. Just dump the whole thing.

Recomposition

However you approach editing the song, please don't think that
editing is a mechanical process, or something to just 'get out of the
way'. Think of editing as a process of 'recomposition'. The end result
should sound, to a new listener, like the song was originally written
that way. And if your edit is really good, it might be even better than
the original recording!

A Record-Producer.com reader is importing 16-bit audio into a
24-bit session. He is concerned whether there will be any quality loss.

Here's a protip for life in general - worry about the
things that make a difference, don't worry about things that don't.
Another - You'll die if you worry, and die if you don't. So just get on
with life and stop worrying.

We get a lot of inquiries here at RP
Towers from people with worries. The usual worry is that they don't have
the right equipment. Sometimes they are justified - plugging a
microphone into an interface with only line inputs is never going to
lead to success. But no-one knows these things automatically so we are
happy to help.

But quite often we find that people are worrying
about things that don't really make any difference. We take it that
anyone who follows RecordProducer.com and Audio Masterclass is primarily
interested in making music and/or recordings of a professional
standard. And by 'professional standard' we mean a quality that will
satisfy a typical industry client, or sell into the market.

So let's have a look at the 16-bit to 24-bit conversion issue. Is it an issue at all really?

The first thing to consider is the significance
of each of the bits. Those sixteen bits are used to quantify the
voltage level of an analog signal in digital terms, in 65,536 steps (2
to the power 16).

So if at any instant the digital signal is at
1001010001101011 then that is a digital description of the instantaneous
level of an analog voltage. In digital terms, the left-most 1 or 0 is
the most significant bit. The right-most 1 or 0 is the least significant bit.
Going from left to right, the signal is described with greater and
greater accuracy, or you could say in finer and finer detail.

Sixteen
bits are sufficient to describe a signal with a dynamic range of 96
decibels from its loudest to its quietest parts. (In real life it's a
little less, but we'll stick with the theory.)
96 decibels is a
HUGE dynamic range. Play some music really loud through your monitors.
Now reduce the level by 96 dB. What can you hear? Nothing. Or as near to
nothing as makes hardly any difference.

When a 16-bit signal is
imported into a 24-bit session, all that happens is that eight more bits
are added to the right of the previously least significant bit. In this
case, they should all be zeroes as the 16-bit signal has no information
here.

But what about dither?

Ah, there just has to be a complication....

One
problem with digital audio is quantization distortion at very low
levels. It sounds nasty and it is something to be concerned about. The
way it is dealt with is to add a little dither noise right at
the end of the recording chain. This eliminates the distortion and
although adding noise sounds like a bad thing, it isn't - it's totally
good. You will almost certainly have a plug-in for it.

So what difference would it make whether or not your 16-bit files were dithered?

Well
if they are undithered what will happen is that quantization distortion
will remain in the lower levels of the 16-bit signal.

If your
16-bit files are dithered, then all should be fine. Unless of course
your DAW's import function adds dither without you knowing about it, in
which case you have dithered twice so your signal is now noisier than it
ought to have been.

The question is of course, do you know
whether your 16-bit files are dithered? If they sound free from
distortion at very low levels, they probably are. And if you can't hear
any quantization distortion, well it doesn't really make much
difference.

Dither is something that people can often get really
worried about. Ideally you wouldn't add any yourself at all and leave it
to the mastering engineer.

But why not explore dither for
yourself - play around with some very low level audio -70 or -80 and
below. Switch dither in and out; try different types of dither. When you
hear what it does for yourself, you will be in a good position to make
decisions.

In summary, if you need to import 16-bit files into a
24-bit session, just go ahead and do it. Listen closely - if they sound
like they need dither then add it (at the 16-bit level). If they don't,
then just leave them be. Why worry?

Thursday, December 20, 2012

No Limit Sound Productions is pleased to announce the release a of newinstrumental music album

by Jordan.

Instrumental New Age music aids in stress management."Solace"
album by Jordan takes you to an added dimension. Solace is defined as a
source of relief and cheer. In the stress of your daily life you can
relax with this New Age Instrumental Music.

Q: "Could you tell me please whether the equalizer should go
first, then the limiter. Or should the limiter go first?"

Whether an equalizer should be placed before or after a limiter depends on the purpose you are using the limiter for.

Sometimes
a limiter is used as an extreme form of compression, for musical
purposes. In this case you can place the EQ either before or after the
limiter, depending on what you want to hear. Try it both ways and choose
whichever you prefer.

However limiters are more normally used
when you don't want the signal to exceed a certain level. This would be
the case in mastering, and you would use limiters in live sound and
broadcasting too.

In addition to the above, you may also want the
signal to come up to a predefined level, not merely not to exceed it. In
this case you would set the limiting threshold so that the signal
frequently triggers the limiting action.

Since equalization can
increase the signal level, then it must come before the limiter. If it
comes after the limiter, then the signal may go higher than the level
you have set.

You may be using EQ to reduce certain bands of
frequencies, in which case the signal level will be lowered. Even so,
the equalizer should come before the limiter, otherwise there is little
point in the limiter being there in the first place.

So, for musical purposes place the EQ where you like. For signal control purposes, the equalizer should go before the limiter.

Wednesday, December 19, 2012

Capacitor microphones used to come each with their own power
supply. Then phantom power was invented so that any number of mics can
be powered from the mixing console. So why are some manufacturers
returning to the old ways?

If you have a very long memory, you will remember
microphones such as the AKG C12 and Neumann U67 - tube microphones that
came with their own dedicated power supplies. You plugged the mic into
the power supply, the power supply into the mains, and took a feed of
the mic signal via the power supply to the mixing console.

Doing this with one or two mics isn't any big deal. But what if you
wanted to have twenty mics on an orchestral session? You would soon get
tired of all that extra plugging.

But somewhere in Norway, way back in 1966, plans were afoot to change all of this.

Norwegian
State Television at the time was already using a 48 V DV powering
system for much of its equipment. They wanted to take advantage of this
ready supply of voltage to power their capacitor microphones, rather
than use a separate power supply for each one.

The Neumann company
responded to this with their model KM 84. This is a small FET
(field-effect transistor) microphone and was the first to be capable of
being phantom powered.

Neumann's phantom power system uses a 48 V
DC source and sends this equally to the hot and cold conductors of the
audio signal cable, and thence to the microphone. The word 'phantom' is
appropriate because you don't see the power supply - it can be contained
within the mixing console and requires no additional wiring beyond the
signal cable.

The word phantom is also appropriate because unless a
microphone is designed to receive such power, there is no voltage
difference between the two signal conductors. A microphone that is not
designed to use phantom power simply will not notice it is there (as
long as it has the usual output transformer).

Phantom power is
supplied to the conductors through two well-matched 6800 ohm resistors.
In fact one power source can be used, and voltage supplied to many
microphones, each through a pair of resistors. This provides protection
against short circuits - even if one mic is shorted to ground, only 14
milliamps can flow, which is peanuts to any properly designed power
supply.

Once phantom power was accepted by the microphone
manufacturing industry, the limitation on the number of microphones
employed caused by all those individual power supplies was removed.

The
strange fact is that currently there is an increasing number of
microphones that are returning to the old ways. In the case of tube
microphones, this can be justified - tubes require higher voltages.
Also, microphone that are capable of handling very high sound pressure
levels can benefit from a higher power supply voltage.

However in
the vast majority of cases, phantom power works just fine. It is in fact
a brilliant invention - brilliant in its simplicity.

A RecordProducer.com reader, like so many of us, finds it
difficult to get into the groove. Is groove a lost art?

By John Speed

John Speed of Montreal added this comment to our recent discussion on groove,
offering additional insight on just how hard it is to 'get into'.
Perhaps sequencers and quantization have taken too much of our attention
recently...

"I think this is the first time I have heard someone
really try to talk about the issue of groove. Lots of musicians say the
word but few know how to get there.

I have been working with two
professional musicians, bass and percussion, for three years now. The
bassist is about 90% technician, a hard worker, perfectionist. We work
hours on counting beats and bars and trying to hit the click track or,
more so, make it disappear.

I am an intuitive singer songwriter,
not a great player but competent. I have learned a lot from this work
and my playing has improved but the band doesn't really "work"!!

Over
three years of practice and gigs I think we have hit the groove for
maybe a total of 3 minutes of playing time. When I play alone I hit it
often. I recognize it immediately when it happens but it comes seemingly
by chance and we do not seem to know how to get there by design, even
though we often discuss the fact that we need to find this magic if the
music is really to become acceptable. We owe this to our audience.

I
tell myself to just keep working, it takes work, lots of work, and this
is the only way there. I would like to hear from others on this
subject, I feel like I am missing something very important. Many thanks
for your good work."

Everyone should build a loudspeaker at least once in their
life. But for this would-be loudspeaker builder, their first attempt was
something of a disaster...

There are some things that you just have to do at
least once in your life - for example see a total eclipse, perform a
parachute jump, and make your own set of loudspeakers. Actually, I might
give that parachute jump a miss. But I've done the total eclipse. I
remember the immense sense of collective disappointment of the thousands
of would-be viewers gathered together at Parc l'Eclipse (really!) near
Cherbourg when a cloud obscured the event from view at the last moment. I
can also remember the massive sense of disappointment when I heard what
my first self-built loudspeaker sounded like.

It
was back in the hazy days of the 1970s, when the rule book of
loudspeaker design was still some way from completion. It was then the
fashion for guitarists to have a 'stack' comprising an amplifier and two
4 x 12 loudspeaker cabinets. I would have liked to have a Marshall
stack, just like Jimi, but I couldn't afford one. So I bought eight
12-inch drive units, the cheapest I could find (far from the quality of
the Celestion illustrated!), and several sheets of chipboard. I bought
the chipboard because it was the cheapest material I could get, but I
later learned that it is actually quite a good material for loudspeaker
cabinets.

So I sketched out a design for a cabinet that was rather
larger than the Marshall equivalent. I don't know why I did that... yes
I do ' I just wanted my stack to be bigger than anyone else's! Then
over a couple of evenings I put that first cabinet together, covered it
with a cheap Rexine imitation, fitted the drive units and handles.
Finially I wired up the drive units to the jack socket and screwed on
the back. I have to say that it looked great. Of course I couldn't wait
to hear it, so I plugged in my amp and guitar and performed my best Pete
Townsend power chord.

Er... there was something
wrong. It didn't sound good at all. All the drive units were working but
the sound was just wrong. Over the next few days I came to the
conclusion that I didn't know as much about loudspeakers as I needed to,
so I decided to cut my losses and sell the cabinet. So I advertised it
at a price that just about covered the cost of the materials. I soon got
an enquiry from a local working mens' club, as they were called in
those days. Since I'm an honest trader (with 100% positive feedback on
eBay, as of writing!) I needed to give them a demonstration so that they
could properly consider what they were buying - a speaker that worked
but wasn't all that good. So I took the cabinet round to the club and
played some music through it. The committee were satisfied and said they
would have the pair. "The pair?", I don't know how that happened, but
since I had the parts for the other speaker, I realized that if I wanted
to get my money back, I would have to make another one.

So
we shook hands, although they did take the opportunity of exploiting my
youth and innocence and knocked down the price to about two-thirds of
cost. I set to work building the other speaker. Once finished, I had to
test it of course. And... it sounded great! It was exactly what I wanted
a 4 x 12 cabinet to sound like, and I reckon it would have given the
Marshall a run for its money. So I looked at the first speaker again to
see what had gone wrong. I spotted the error straight away, when hours
of looking hadn't helped just a few days ago. I had wired the four drive
units in the conventional series-parallel way, but two of the drivers
were wired in reverse phase. So effectively at any instant when two of
the drivers were pushing at the air, the other two were pulling. I
corrected my error, and I now at last I had my brilliant twin 4 x 12
stack. Except I had agreed to the sale and later that day the club sent a
van round. I never did make any more 4 x 12 cabinets, but at least I
knew that I could if I wanted to.

In contrast to my London Olympics tickets that cost me
upwards of £150 each and were the cheapest I could get my hands on, my
ticket to Roland's V-Piano Grand UK premiere in the Britten Theatre of
the Royal College of Music was free. Now that's an offer I couldn't
possibly refuse!

I imagine Roland must have my address from the registration card I
sent in when I bought my own V-Piano, in its original stage version.

But why would I buy a digital piano? I hate digital pianos. All of
them, without exception. I can play a conventional piano for personal
pleasure for hours. I even enjoy practising scales (some people enjoy
working out at the gym, so why not?)

But a digital piano - well there just isn't any enjoyment to be
had. OK, some of them do make a noise of reasonable quality. But they
don't feel good to play.

But the one thing that digital pianos do have going for them is
practicality. They are more compact, don't need tuning, and you can use
them with a MIDI sequencer.

Digital pianos are also easier to record. Of course it is perfectly
possible to get a great recording of a good-quality acoustic piano,
well maintained and tuned, in a good studio or concert hall. But try
doing the same thing with your upright at home. Suddenly recording just
became a lot harder.

So although I love playing my Yamaha acoustic pianos (plural, but
not at the same time), I wanted a digital piano for recording. I tried
them all and bought the best - the Roland V-Piano.

Stage piano to grand piano

Roland describe the V-Piano as a 'stage piano'. It's the kind of
thing you would play with a band, like you would once have played a
Fender Rhodes.

Turning it into a grand piano however raises a big question...

Why?

The standard V-Piano has all of the advantages of digital pianos I
listed earlier, which to summarize boil down to practicality. But a
digital grand piano lacks the advantage of compactness and portability.
And although a digital grand piano may be easy to record through its
line out sockets, a conventional grand piano is easy to record too, in a
decent acoustic space.

So, before the event, I wondered to myself what the point of the
V-Piano Grand could be. The only answer I could come up with was that I
expected it to be superior to a conventional grand in some way.

If you're not familiar with the V-Piano, now is the time to learn
that it works by modeling, not samples. And it can model grand pianos,
upright pianos, antique pianos. And pianos that don't exist - like
pianos with three strings for every note, pianos with silver strings,
pianos with a glass soundboard.

So there is some potential here for the V-Piano to be better than a conventional grand. But is that potential fulfilled?

The V-Piano Grand in concert

On seeing the V-Piano Grand in real life for the first time, I
found it smaller than I expected. On a concert stage, you expect a piano
to be of a certain size. Compared with a Steinway Model D, this was
pint-size. OK, quart-size.

Pianist Daniel Tong took the stage and grasped the keys in a masterly fashion. And the sound that came out...

Well it sounded like a piano. But I have to say that although the
digital modeling of the V-Piano is wonderful, it sounded like a piano
played through loudspeakers. I don't want to over-emphasize this point
because it was only a little 'speakery', and for many purposes this
would pass unnoticed. I am sure that in a blind test, many listeners
would not be able to tell. But in terms of sound quality, there is no
way that the V-Piano Grand is better than a conventional piano.
Steinway's business model is safe, for the moment. However...

Have a go

At the end of the concert I hung around a little until most of the
audience had departed. I and a few others then found an opportunity to
hop up on stage and have a go on the V-Piano Grand for ourselves.

First up was a pianist of excellent ability, which gave me the
opportunity to walk around the piano, as I would if I were selecting a
mic position for a conventional piano. I expected the sound to be
localized from the loudspeakers (you can see the grilles), but no, the
sound was very full and appeared to come from the whole of the
instrument.

I sidled up to the piano stool and took the next spot. I was surprised - the V-Piano Grand is very
pleasant to play. This for me is the sticking point for all other
digital pianos. They are not nice to play. But the V-Piano Grand plays
very much like a conventional piano; the sound is alive and responsive.
In terms of playing for pleasure, the V-Piano Grand could make an
alternative to a conventional piano. It can't match a large,
high-quality, big-name grand perhaps, but it's a contender against less
highly-specified models.

Hotel lounge piano?

Having had my go on the ivories, I asked an onlooker what his
interest in the V-Piano Grand was. It turned out he was a hotelier and
he was interested in having one for his lounge. I'm not so sure that is
what Roland had in mind, but a sale is a sale, so if they can convince
him on the grounds of practicality, I'm sure he will be pleased.

Finally I had a chat with one of the Roland guys, who made
everything clear for me... The purpose of the V-Piano Grand is to be a
flagship product. Like car makers who produce a really high-end model in
limited quantities. They don't expect to make much of a profit from
their ultra-sporty or ultra-luxurious models, but the publicity they can
get from them is invaluable, and the new technologies they develop can
trickle down to their standard range of products.

Digital pianos are big business for Roland and they have an
incredible variety of models - far more than you would expect unless you
took a look at the catalog.

So in the future we can expect to see V-Piano technology trickling
down to their more affordable models. I'm all for this - the digital
modeling in the V-Piano is fantastic. The V-Piano Grand might not knock
Steinway off its perch, but it does sound good and brings excellent
playability to the digital piano market.

I don't suppose I'm going to buy a V-Piano Grand, but I'm very
happy with my V-Piano (non-grand). In fact I think I'll go and practise
some scales on it right now...

A Record-Producer.com visitor asks whether a good
analog-to-digital converter is necessary. Or will any old converter do?

"Do you have to use a good converter for a microphone to compete with the industry?"

Firstly,
a little Level 1 explanation for newcomers to recording... A microphone
needs a preamplifier to bring the level up from a few tens of
thousandths of volts to around one volt. Then the signal goes through an
analog-to-digital converter so it can be input into a digital audio
workstation.

So firstly you have to use a preamplifier that is
comparable with those used in the pro industry. That's another matter
entirely, so I'll assume that this is already taken care of.

So, do you need a good analog-to-digital converter, which I'll call a converter for short, or will any old converter do?

Well,
without doubt it would be nice to have the best converter in the world,
that was comparable with the very best that industry pros use. There is
no doubt that it is always the right thing to do to aspire to the
ultimate standard available.

But what if you can't afford the best? Will your recordings be ruined?

One
way of looking at this is to go back into the history of digital audio,
back to the early 1980s. The converters they had then were primitive
compared to what we use now.

Sometimes they were not even
'monotonic', meaning that an increase in voltage didn't always result in
an increase in the digital numbers that came out.

Even so, many great recordings were made with such converters. Recordings that stand the test of time now.

I
would venture to bet that even the worst converter that is sold into
the pro audio market these days is by far superior to the best that was
available then.

So, according to this logic, you don't
particularly need to worry about the converter. Yes, buy the best you
can afford and couple it with a good preamp. Then forget about it and
work on your music, your studio acoustics, your microphone technique
and your mixing skills. They will make infinitely more difference than
anything else.

One of the potential problems of compression is pumping. And in this example it's about as bad as it gets.

There are many good features about this video, and it is extremely useful for its intended purpose.
But the audio has a problem. Two problems in fact.

The first is the noise. Clearly the audio is being recorded direct into
the camcorder through its built-in mic. If it is a tape-based camcorder,
then this could be the reason for the noise.

But also, there is a huge amount of pumping, due to the automatic gain control (AGC) of the camcorder.
When the player hits a note or chord, the AGC kicks in and quickly
lowers the gain. But as the notes decay, the AGC relaxes and allows the
gain to go up again. The result is that, between notes, the volume
swells.

The solution of course is to record the audio separately onto a
dedicated audio recording device, then sync it again in a video editing
app. Or, if possible, use an external microphone and set the camcorder's
gain manually, switching off the AGC.

Of course, either way is more fiddly than straightforward
point-and-shoot. Although the results may be better, it might result in
fewer videos from this YouTube contributor. Some might say that fewer
but better videos would be the way to go. On the other hand, this
pianist has set himself a mighty task of recording a lot of music that
many people will undoubtedly find useful.

Friday, December 14, 2012

An RP reader wants to capture his hihat in high fidelity. So what is the right mic for the job?

If the question is, "What is the right mic for hihats?",
then another question could be, "Right for what?" Everything is
subjective in the listening experience, and ultimately it's all about
opinions. Or rather, it's about what sells.

So let's start with the textbook answer...

The
hihat falls into the class of instrument known as metallic percussion.
Bash two pieces of metal together and you are going to get a lot of high
frequencies coming out, which you will want to capture accurately.

High
frequency sound causes the diaphragm of the microphone to vibrate back
and forth rapidly, so you need a mic with a diaphragm that is capable of
vibrating very easily at a quick rate.

That therefore rules out
the dynamic mic. Dynamic mics have a coil of wire attached to the
diaphragm, which makes the diaphragm heavy giving it a certain amount of
inertia. Although dynamic mics often have a useful 'presence peak',
they are not renowned for crystal clarity at high frequency.

So it's going to be a capacitor mic then. Large or small diaphragm? Tube or transistor?

Let's
start with tube or transistor first. The reason for the continued
existence of tube microphones is the 'thickened' sound they produce. Who
wants a thick hihat? Strike that one.

Now, large diaphragm or
small? Well with the large diaphragm we are once again in inertia
territory, and it is often thought that the resonance of a large
diaphragm, even if well controlled, can smear high frequencies.

So,
going through the possibilities, we come to the textbook answer that we
should use a small diaphragm capacitor microphone on the hihat, which
will have a transistor internal amplifier because that's all that's on
the market these days (unless someone knows different?)

But that's the textbook answer. What's the real-world answer?

Well
I had the experience a while back of running out of microphones. There
were not enough small-diaphragm capacitor microphones to go round and
something had to give. What is the instrument, I thought, that least
matters if it has the wrong mic?

Aha - the hihat!

So that is
where I made my compromise. I can't remember where the small diaphragm
capacitor mic went, but on the hihat was a Shure SM58!

Now for the
hihat, this is about as un-textbook as you can get. But you know what?
It didn't matter. In the mix, no-one would have cared what mic the hihat
had, although to be fair it was a rock music arrangement. With other
styles of music it might have been a different issue.

Has anyone else had success using the 'wrong' microphone? Or did the 'right' mic ever not perform as expected?

I always set up my projects at 24 bits/44.1 kHz, but
recently I noticed how sample libraries are offering 32 bits/96 kHz. Do
you think we have to use them?

The answer to this question depends on whether you use sample libraries or create them.

If
you are a user of sample libraries then 24/96 is good enough for any
purpose. 24/44.1 will sound perfectly OK to 99.9% of potential listeners
and the other 0.1% probably wouldn't know unless you told them.

There
is however an area where 24/96 can be audibly better than 24/44.1, and
that is where samples are used transposed down from their normal pitch.
The higher sampling rate will help preserve higher frequencies.

It
is worth saying however that sometimes the artefacts of sampling are
exactly what is wanted. If, for example, you wanted to achieve a retro
80s sound.

In theory a 24-bit sample should offer a dynamic range
of 144 decibels, which is wider than the ear can cope with. A 32-bit
sample could in theory extend this to 192 dB. Bigger numbers are always
better but, once again, few would actually hear the benefit.

If
you produce sample libraries, then it is best to go for the most
excellent recording quality possible. That way you will keep up with, or
ahead of, the competition. You will win in the numbers game and
safeguard yourself as much as possible against further advances in
technology.

Some microphone and instrument preamplifiers have a variable
impedance selector. What kind of difference will it make to your sound?
What are you missing if you don't have it?

Why your preamp should have an impedance selector. You're missing out if it doesn't...

Some
microphone and instrument preamplifiers have a variable impedance
selector. What kind of difference will it make to your sound? What are
you missing if you don't have it?

Here is an excellent example of a
preamplifier with a variable impedance selector, the Little Labs Multi Z
PIP. Before I go further, let me tell you that it costs around $600 - I
wouldn't want you building up a desire for one and then finding out you
can't afford it!

Yes, it's a glorified DI box, but what glory...
there is a level control, which is a bonus compared to most DI boxes,
but also there is this all-important variable impedance selector, known
here as 'Input Circuit Select'.

What does it do?

To put it simply,
if the switch is set to 'hi Z', then the input impedance is high and
the unit draws hardly any electric current from the sound source, which
I'll assume is a standard electric guitar.

If the switch is set to 'lo Z', then the unit will attempt to draw a large current from the pickups of the guitar.

Now
where the difference arises is in the ability of the pickups to supply
current. A guitar pickup isn't very good at providing current, so where
the hi Z position isn't asking an awful lot, and the pickup is quite
comfortably able, the lo Z position demands rather more current than the
pickup can successfully supply.

So what happens? Well imagine if
the pickup was shorted out by a piece of wire. In this case the Z would
be so low as to be zero. In this situation, the pickup will provide no
voltage, hence there is nothing to amplify. So a low Z, but not zero,
will lower the output voltage.

That doesn't sound good, does it?
But what happens in reality is that the pickup is more capable of
supplying current at certain frequencies, according to its design, than
others. So the loss in voltage is frequency selective.

The hi Z
setting, captures the full range of frequencies of the guitar, the mid Z
and lo Z positions will capture different frequency responses, in
general with less high frequency energy.

Yes, the impedance
selector is a kind of EQ control. But it's an EQ that is very dependent
on the characteristics of the pickup, and every pickup will react
differently.

So being able to select different impedances makes
the instrument/preamp combination a kind of a symbiosis, where one
reacts to the other to create a unique sonic character - you can't get
this with EQ alone.

So with a guitar such as an old-style
Stratocaster with a three-position selector switch, the three impedance
positions on the Little Labs Multi Z PIP allow a total of nine sound
combinations.

The pinna of the ear helps to collect sound, obviously. But
why does it have those complicated folds? And why do we only have two
ears and not three?

For holding your sunglasses on would be one answer. Helping locate, or localize,
the direction of a sound source is another. Much of the design of the
human body is down to the requirements of living in the wild on the
plains of Africa tens of thousands of years ago.

In those days
there were priorities other than getting a deal with a record label.
First and foremost was personal survival, closely followed by the
survival of the species. So on any encounter with something new and
unfamiliar there were three possible courses of action... eat it, escape
from it, or mate with it!

The ear is adapted by evolution to help
us survive, by pinpointing sources of danger, and by helping us
communicate. So the pinna, which is the proper term for the ear 'lug',
helps collect and localize sounds. The shape of the pinna, like a
miniature satellite dish, collects sound over a relatively large area
and funnels it into the auditory canal to the ear drum. At the same
time, the body of air it partially encloses resonates at frequencies
around 3.5 kHz, thus acoustically 'amplifying' these frequencies.

This helps us hear speech more effectively - not the fundamental tones but the sounds that mark the differences between words.

The
main localization function is provided by the fact that we have two
ears. But the pinna also helps. By being forward-facing it helps us
differentiate between front and back, which the mere fact of having two
ears does not. Also, the complex folds cause reflections that subtly
boost and cut different frequencies according to the height of a sound
source.

So although we do not have any specific detector for the
height of a sound source, we can in fact localize quite well in the
vertical dimension. So you can tell instantly whether that strange sound
you hear is a rat nibbling at your boot laces, or a bird about to dive
bomb you with its payload.

Which reminds me of a joke. Did you
know that Captain James T. Kirk of the Starship Enterprise did in fact
have three ears? His left ear, his right ear, and his final front ear!
(Final frontier -- get it??)

I have heard that mixing consoles have busses, but I can't
see anywhere I can control or operate a buss. Could you tell me please
what a buss is?

In audio we use the words 'bus' and 'buss'. Either spelling
is acceptable, however for this answer we will use 'bus' to mean the
road vehicle; 'buss' to mean the component of a mixing console that is
under discussion.

Think of a road bus at the outer end of its
route in the suburbs of a city. Initially it is empty, but as it wends
its way to the city's heart, it picks up passengers at every stop.
Eventually it gets to its destination and everyone gets off.

A
mixing console buss is similar (in an analog console). It is a wire or
metal rod that starts at Channel 1 at the left of the console and ends
up at a group output or master output on the right. Let's say that we
are considering the buss for Group Output 1.

As it traverses the
console, it picks up signals from any channels that are routed to that
buss, just like the road bus picks up passengers.

When the buss arrives at Group 1, it delivers all of its signals.

Of
course, you can only take an analogy so far. The buss in a mixing
console doesn't move, and it doesn't return anything back to the
channels (there is no reverse direction to its route).

An analog
mixing console has one buss per group, plus one buss for each of the
master outputs, plus one buss for each auxiliary send that it has.

The video is introduced by explaining what a pop filter is
and why a person who records podcasts should build one. At 1:30 the
instructor introduces the materials needed to make the filter. At 2:00
he begins instructing viewers on how to make the microphone filter
beginning with the treatment of the nylon...

By YouTube Genie, video courtesy YouTube

The video is introduced by explaining what a pop filter is
and why a person who records podcasts should build one. At 1:30 the
instructor introduces the materials needed to make the filter. At 2:00
he begins instructing viewers on how to make the microphone filter
beginning with the treatment of the nylon.

At 4:00, the instructor shows viewers how to assemble the nylon and
the embroidery hoop. He cautions people who are working alone that they
must be clever in stretching the nylon tightly over the hoop without an
extra pair of hands for help.
At 8:05, once the filter is in place, the instructor begins to talk
about the quality of recording vocals.

If a person speaks nearer to the microphone, his or her voice sounds
deeper. When speaking further from the microphone more of the high notes
are caught by that transmitted resulting in a higher pitched sound. Ultimately, the filter is able to prevent the hisses and pops that occur
with a naked microphone.
Often times, though the speaker is unaware of a problem when he or she
listens back to what has been recorded, every p, t and s sound stands
out with a distinct popping or hissing.

Expensive recording tools can prevent this problem, but if the
podcast is only a hobby, the recorder might not want to spend too much
money on a solution to the popping. To make a microphone pop filter for
under ten dollars, all that is needed is a four-inch, wooden, embroidery
hoop, a one foot length of slip tubing, a ten inch by two inch moldable
gauge, a ratcheting zip tie a single black nylon, clip pliers, scissors
and a screw driver.

This cheap solution will resolve every pop and hiss making the final
recorded podcast sound professionally made. The assembly of the filter
takes under ten minutes. To begin, the builder should take the nylon and
cutting the toe away, then the leg. The remaining nylon tube should be
cut lengthwise and laid flat. The length of the nylon will be roughly
twice the breadth. Folding it once over will make a square and provide a
double barrier.

The builder should then take the nylon, folded in half and stretch it
over the small round of the embroidery hoop. Once it is stretched
tight, the larger round of the hoop should be slipped over the nylon on
the small round and tightened with the screw fastener. After the hoop is
assembled with the nylon stretched tightly over it, the excess fabric
should be cut away.
Taking the slip tubing and fastening it to the base of the microphone
stand, the builder provides an anchor point for the hoop.

The tubing is best positioned roughly two inches from the microphone
at a obtuse angle to the mesh. Once the tubing is in place, the hoop can
be fastened to the area just below the microphone using the zip tie.
The zip tie can securely join with the tubing by lining up the screw
fastener and end of the tube. The excess on the zip tie should be cut
away. For a professional appearance the moldable gauge can be slipped
over the tubing to make a uniform black pop filter, once assembled a
podcast can be recorded free of hissing and popping sounds.

Whoever designs microphones clearly thinks that they all
should be microphone-shaped. But isn't it about time we had something
more appropriate to the way we use them?

OK, I'll tell you the image
that's in your mind. It's either a thin near-cylindrical end-address
microphone, or a fat near-cylindrical side-address mic.

You're thinking of a Shure SM57 or a Neumann U87!

Well, something like that.

The
mics we use today are clearly descended from those designed in the
1940s and 1950s. And the classic shapes of the 'pencil mic' and 'bottle
mic' are without doubt the most popular. Alternatives are very thin on
the studio floor, or live stage.

I was watching some old concert footage on TV the other day and I noticed the mics on the guitar cabs.

They were clearly the Shure SM57 model, and they were simply hanging down in front of the cabs by their cables.

Of course, this can be seen as the lazy man's way to mic a cab. Or perhaps there weren't enough stands available.

But clearly when used in this way, an end-address mic is pointing in the wrong direction.

The problem wouldn't arise if we had microphones that were specifically designed for guitar cab miking.

If,
for instance, a mic could be made in a rectangular shape with a
side-address diaphragm, it could dangle from the cable with the
diaphragm pointing at the speaker cone.

Or perhaps a new type of
stand could be devised to fit onto the cabinet, rather than taking up
floor space. Surely that would be better than using a conventional
stand.

Perhaps the mic and 'cab-stand' could be one integrated structure.

Sometimes I wonder whether we are too set in our ways and we need a few 'crazy' ideas to refresh the process of making music.

If you have a studio business, doubtless you want it to
continue for decades to come. But if you don't understand the client
life cycle then you'll soon be in trouble.

If you have a recording studio business, then
congratulations to you! It's a hard trick to pull off successfully when
everyone has a pro-quality DAW at home.

But there are some types
of recording that you really do need a professional environment for,
with soundproofing and acoustic treatment too.

Having a studio is not the same as keeping a studio. Many businesses, of all kinds, do not fully appreciate the client life cycle.

When
your business is new, you will spend a lot of time attracting clients.
Some of those clients will be one-offs, others will come to appreciate
the quality of what you do and come back again and again.

Gradually you will acquire more and more regular clients and they will provide most of your income.

You
might come to view one-off clients as a bit of a nuisance. They don't
understand how you work, they ask for unusual things, you're not sure
whether they will pay on time etc.

Eventually you might find yourself working only with your group of regular clients.

Now let's look at things from the client's point of view...

Every
client is a new client at the beginning of their relationship with your
business. This is the 'birth' end of the client life cycle.

Eventually
the client will become mature and will use your services again and
again. It's win-win because you get regular income and the client gets a
service of known and repeatable quality.

But eventually the
client will 'age'. They might outgrow the service you can provide and be
forced to go elsewhere. They might find a cheaper solution. They might
go broke or otherwise go out of business. They might actually retire.
This is the 'death' part of the client life cycle.

So if your
business relies on a group of regular clients in the mature phase of
their life cycle, you cannot expect this to go on forever. One by one
they will drop off and die.

And now you are faced with the
unfamiliar situation of having to attract new clients - something you
perhaps have not done for years.

The solution, for any business,
is never to rely on regular custom. You have to attract new clients all
the time. There should be a regular cycle of birth-maturity-death among
clients and you should expect and welcome it.

Many studio
businesses have failed over the last ten years or so. In many cases this
could be because they haven't understood the client life cycle
properly.

Do your recordings sound natural? Or do they sound
'microphony', electronic or digitally processed? How can you tell?

The other day, I found myself advising someone that their
recording of speech was good but it didn't sound natural. I further
advised that the sound quality they had achieved was commonly heard on
the radio, but it would be tiring to listen to for a long period, if the
recording was part of an audio book for example.

It's worth
reflecting for a moment on what we would consider naturalness, in a
recording, to be. Fortunately we have examples of natural sound around
us all the time, so there is plenty of material for comparison.

Perhaps
the most useful natural sound is the human voice. Our ears are very
closely attuned to the sound of the voice and we hear it all the time,
and - most importantly - pay close attention to it. Of course I do mean
the human voice as produced from a human larynx, throat and mouth,
traveling directly through the air to your ears, not via a loudspeaker.

Let's consider therefore how we can compare the natural human voice with the sound of the voice reproduced via a loudspeaker.

Firstly,
the person we choose to provide our hypothetical example of human
speech should have a reasonably normal quality of voice. Professor
Stephen Hawking writes excellent books on cosmology, but he isn't going
to make a good example. Neither would a 40-a-day smoker. But we don't
have to be too choosy. Apart from a few wayward examples, almost anyone
would do.

Now we have to consider context. Should we consider the
example of a friend spotting you from the other side of a busy road and
shouting you a greeting? Well we could, but it doesn't have a lot of
commonality with anything we would be likely to do in audio.

What
about a lover whispering sweet nothings into your ear? That might be a
desirable scenario, but the sound of the voice at extremely close range
is difficult to mimic accurately. It's an interesting challenge, but we
need something simpler.

So what about someone talking to you in a normal voice from a distance of two meters? That's just over six US feet.

This
is a good test because it is a commonplace situation with which we are
all very familiar. Also, it is practical to simulate with audio
equipment. Bear in mind that most loudspeakers have at least two drive
units and a certain amount of distance is required to allow the sound to
integrate. A distance of one meter wouldn't be enough as small changes
in listening position produce significant changes in perceived sound
quality (and that is something to consider when using near-field
monitors).

So imagine this... There is a visually opaque but
acoustically transparent curtain in front of you, behind which there is a
person, ready to speak from a prepared script (or you could do it in
the dark). And also there is a loudspeaker, mounted with its central
axis at the same height as your volunteer's mouth and as close as
possible to one side. Through this loudspeaker will be played a
recording that this same person made earlier. An assistant has
previously checked that everything is working and that the levels are
very similar.

So now you hear a voice. Is it human or is it the
loudspeaker? Now you hear another voice. Or is it another voice? Is it
perhaps the same sound source? Or has the source changed but there is so
little difference that you can't tell?

You could carry out this
experiment for real. Or you could consider it to be a test of
naturalness in audio, and have this thought in your mind next time you
need a recording to sound natural. Listen to your recording and ask
yourself whether you would be fooled.

Although the human voice is
the supreme test of naturalness in audio, it is also worth considering
whether your recordings of acoustic instruments, including drums, sound
natural. And if they don't sound natural, should you be trying to get
closer to a natural sound, or are you trying to improve on nature?

Of
course, naturalness isn't always the requirement. But it is a very
useful benchmark of audio quality. Listen to your recordings closely and
ask yourself which aspects don't sound natural. And whatever doesn't
sound natural, ask yourself whether it is a defect, or an improvement.

My drums and bass guitar sound loud and heavy when I play
them, but they don't sound heavy enough when I make a recording. How can
I make them heavier?

Ah... you're suffering from lightweight drums and bass syndrome! This is very common in recording.

Let's look at the bass guitar because this is more likely to be suffering from this problem.

The
principle cause of LBGS (lightweight bass guitar syndrome) is the way
the ear perceives sound. When you plug your bass guitar into an
amplifier and speaker and turn up the volume, your ear doesn't only
perceive loudness, it interprets that loudness as heaviness too.

The microphone doesn't. It picks up the sound exactly as it is - merely loud.

When
you set the gain correctly on the preamp, the resulting recording will
be hardly any heavier than if you had set the volume control on the amp
to 1.

Indeed, it might have no apparent heaviness at all.

So what's the cure?

The
solution to this problem is to recognize that loud sound stresses the
ear and causes distortion in the hearing process. So in other words,
you're not hearing the sound as it actually exists, you are hearing your
ears' interpretation of the sound.

Having realized that, you can begin to restore lost heaviness by adding distortion into the process.

This
is best done at source. So if you have a powerful, clean bass amp, you
need to exchange it for one that creates more distortion. So you need
tubes rather than transistors, and low-power rather than high-power, so
that when you turn the amp up, it distorts more.

Your speakers too
might need attention. Speaker technology has 'improved' since the 1960s
and it is possible to design and build drive units that are ultra clean
at high sound levels.

But this doesn't produce a heavy sound in a
recording. It is better to use drive units of 'old school' design where
the cone bends more and produces a more distorted, but heavier, sound.

You
might choose not to use an amplifier and speaker at all, and record
through an amp simulator, or use an amp modeling plug-in.

Amp
modeling is an amazing technique that can mimic the sound of real amps
and speakers. The problem can be that although the modeling seems good,
the result is lightweight.

Here you can improve the sound by
putting the modeled signal through an amp and speaker, which could be
your monitor system with appropriate settings of the controls. Mic this
from a distance of a meter or more so that you pick up some of the
ambience of the room.

You can use this alone, or mix it in with
the original modeled signal. If you mix it, consider time aligning the
two signals so that you don't get cancelation of some frequencies.

This
technique will require a lot of experimentation to get the sound
exactly right. But what you will end up with can potentially blend
clarity and heaviness in exactly the right proportion.